Method for optimizing speech pickup in a communication device

ABSTRACT

A method for optimizing speech pickup in a speakerphone system, wherein the speakerphone system comprises a microphone system placed in a specific configuration, wherein the method comprising receiving acoustic input signals by the microphone system, processing said acoustic input signals by using an algorithm for focusing and steering a selected target sound signal towards a desired direction, and transmitting an output signal based on said processing.

This application is a Continuation of copending application Ser. No.17/381,690, filed on Jul. 21, 2021, which claims priority under 35U.S.C. § 119(a) to Application No. 20187154.8, filed in Europe on Jul.22, 2020, all of which are hereby expressly incorporated by referenceinto the present application.

The present disclosure relates to loudspeaker system. The disclosurerelates specifically to a loudspeaker system comprising a microphonesystem for picking up speech at a distance from the talker, oftenreferred to as a speakerphone. The disclosure relates further to amethod for room optimizing microphone signal performance by mixingvarious DIR filter outputs with e.g. omni and enhanced omni responsesbased on input from reverb estimator.

The description given herein may e.g. be useful in applications such ashandsfree telephone systems, mobile telephones, teleconferencing systems(e.g. speakerphones), headphones, headsets, public address systems,gaming devices, karaoke systems, classroom amplification systems, etc.

The sound quality from any microphone system meant for picking up speechat a distance from the talker is greatly affected by the conditions ofthe room wherein the system is placed. In hard (relatively high amountof reverb) rooms speech intelligibility can be significantly impaired. Amethod to counter this is to make the microphone system directionalprimarily picking up sound from a narrow angle (high directionalityindex) in the direction of the talker thereby excluding some of thereflected sound. The cost of this is complexity and amplified microphonenoise. In very soft rooms (very low reverb time) there will in somecases be no benefit at all from directionality. Using a directionalsystem under these circumstances will only add noise compared to anomnidirectional system and therefore offer worse performance overall.

Simply using a directional system with fixed directionality for allrooms is a suboptimal solution in most situations.

With the above description in mind, then, an aspect of the presentdisclosure is to provide a method and system for optimizing microphonesignal performance based on conditions of a room wherein the system isplaced, which seeks to mitigate, alleviate or eliminate one or more ofthe above-identified deficiencies in the art and advantages singly or incombination.

According to an aspect, a method for optimizing speech pickup in aspeakerphone is provided. The speakerphone comprises a microphone systemplaced in a specific configuration. The term specific pattern is meantas a predefined pattern so that the processing of sound signals from themicrophone system may be performed reliably. The method comprisingreceiving acoustic input signals by the microphone system, processingsaid acoustic input signals by using an algorithm (100) for focusing andsteering a selected target sound pickup direction towards a desireddirection and transmitting an output signal based on said processing. Byfocusing is mean that a directional signal is mixed at a given ratiowith an omnidirectional signal so as to change the degree of focusingthe sound pick up from a sound source external to the device, e.g.speakerphone.

The microphone system preferably provides output signals that eitherthemselves are directional and omnidirectional, or directional andomnidirectional are at least derivable from signals from the microphonesystem. A directional and an omnidirectional output signal may beprovided to the mixing unit and/or a processing unit. The directionalsignal is preferably adaptable so that the directional signal has a mainpickup direction that may be adapted to mainly pickup sound from aspecific direction relative to the housing, e.g. speakerphone housing.The directional signal may be established via two or more individualmicrophones of the microphone system. Further microphones may beutilized. An external microphone system may be connected to aspeakerphone as described herein. Such an external microphone system maybe used in reducing reverberation picked up by the microphone system inthe speakerphone.

Noise reduction may be applied to sound signals picked up by themicrophone system of a speakerphone. Noise reduction may be asignal-channel noise reduction system.

A speakerphone as described herein may be used in conjunction with avideo system e.g. for videoconferencing. Those two components may be incommunication with a computer device or the like, alternatively eitherdevice may include processing and software so as to enable communicationvia a softphone system.

In one aspect, the processing comprises optimizing the target soundsignal while simultaneously rejecting sound signals coming from otherdirections, allowing the pickup of speech from any angle desired,relative to the speakerphone system.

In one aspect, the method comprises the step of analyzing the content ofthe received sound signal in all directions, and automatically selectingthe direction of interest, the target sound signal.

In one aspect, the processing comprises determining a mixing ratio basedon a room characteristic and mixing a directional sound signal with anomnidirectional response signal based on the determined mixing ratio. Aroom characteristic may be obtained by a measurement, e.g. performedusing the microphone system. A room characteristic may be input to thesystem as a setting by a user. A room characteristic may, as disclosedherein, may be determined and updated continuously. At least oneparameter of a room characteristics may relate to possible reverberationlevels in the room in which a speakerphone is placed. A roomcharacteristic may be determined based on a measurement of reverberationby a speakerphone. The room characteristic may be continuously updated.This could be via a measurement by analysis of sound recorded via themicrophone system In one aspect, the method comprising continuouslyupdating the mixing ratio. The processing may be constituted by themixing as described herein. During use the speakerphone is placed in aroom, usually left on a table or the like.

In one aspect, a data processing system comprising a processor andprogram code means for causing the processor to perform at least some ofthe steps of the method disclosed above.

In one aspect, computer program comprising instructions which, when theprogram is executed by a computer, cause the computer to carry out thesteps of the method disclosed above.

According to another aspect, a system for optimizing speech pickup in aspeakerphone of the system, comprising an input unit comprising amicrophone system placed in a specific configuration and adapted forreceiving an audio input signal and providing an electric audio inputsignal, a processing unit adapted for by using an algorithm for focusingand steering a selected target sound pickup direction towards a desireddirection and an output unit for providing an output signal perceivableby a user as sound representative of said electric acoustic input signalor a processed version thereof. In one aspect, the microphone systemcomprises at least six microphones. In one aspect, the microphones arelow-noise digital MEMS microphones are placed in a specificconfiguration optimized to enable speech to be picked up from any angle.

In one aspect, the microphone system is engaged, and the advanced signalprocessing algorithms are applied, and the directional pattern isfocused into a tight beam.

In one aspect, the processing unit is adapted for optimizing the targetsound signal while simultaneously rejecting sound signals coming fromother directions, allowing the pickup of speech from any angle desired.

In one aspect, the processing unit is adapted for analyzing the contentof the received sound signal in all directions and automaticallyselecting the direction of interest, the target sound signal.

In one aspect, the speakerphone system is adapted to automatically steera focused beam to the desired direction for the target signal, even ifthe target sound signal changes position. Steering a directional signalis also referred to as adaptive beamforming.

In one aspect, the processing unit comprises a mixing unit fordetermining a mixing ratio based on the room characteristics and mixinga directional sound signal with an omnidirectional response signal basedon the determined mixing ratio. In one aspect, the mixing ratio iscontinuously updated.

According to another aspect, a speakerphone for optimizing speech pickupin said speakerphone, wherein the speakerphone comprises comprising aninput unit comprising a microphone system placed in a specificconfiguration and adapted for receiving an audio input signal andproviding an electric audio input signal, an input unit for receiving anaudio input signal and providing an electric audio input signal, aprocessing unit adapted for by using an algorithm for focusing andsteering a selected target sound pickup direction towards a desireddirection, and an output unit for providing an output signal perceivableby a user as sound representative of said electric acoustic input signalor a processed version thereof.

A communication device, comprising a loudspeaker system as describedabove in the detailed description, in the drawings and in the claims isfurthermore provided.

An article of manufacture may implement a communication devicecomprising

-   -   a first microphone signal path comprising        -   a microphone unit,        -   a first signal processing unit, and        -   a transmitter unit        -   said units being operationally connected to each other and            configured to transmit a processed signal originating from            an input sound picked up by the microphone, and    -   a second loudspeaker signal path comprising        -   a receiver unit,        -   a second signal processing unit, and        -   a loudspeaker unit operationally        -   said units being operationally connected to each other and            configured to provide an acoustic sound signal originating            from a signal received by the receiver unit.

Thereby a speakerphone or a headset comprising a loudspeaker systemaccording to the present disclosure can be implemented.

The communication device may comprise at least one audio interface to aswitched network and at least one audio interface to an audio deliverydevice. The (one way) audio delivery device is a music player or anyother entertainment device providing an audio signal. The (loudspeakersystem of the) communication device may be configured to enter or be inthe first, full bandwidth mode, when connected via the audio interfaceto a (one way) audio delivery device. The (loudspeaker system of the)communication device may be configured to enter or be in the second,limited bandwidth mode, when connected via the audio interface to acommunication device for establishing a (two way) connection to anothercommunication device via a switched network. Alternatively, such changesof mod can be initiated via a user interface of the communicationdevice.

The communication device may comprise a speakerphone or a mobile (e.g.cellular) telephone (e.g. a SmartPhone) or a headset or a hearing aid.

The communication device may comprise a headphone or a gaming device.

The loudspeaker system may for example be used in a telephoneconference, where the audio gateway may be a laptop, or a smartphoneconnected to an internet network or a telephone network via the firstcommunication link.

Conference calls are a more sustainable approach to collaboration withcolleagues across geographic locations. This means that the quality andreliability of conference solutions is vital to support productivevirtual meetings.

A meeting room speakerphone has two primary functions. It must broadcastremote speech through a loudspeaker, and capture speech in the meetingroom while conveying it to remote participants. Ideally, thespeakerphone must convey speech as intelligibly as possible. This meanssound from the loudspeaker, broadcasted into the meeting room, must beprevented from returning to the remote listener through thespeakerphone's own microphone. This is achieved through a process knownas “echo cancellation”. Quality speakerphones also isolate voices fromboth ambient noise and reverberation, which greatly affect speechintelligibility for remote participants.

Noise and Reverberation

Speech becomes more difficult to understand in a noisy environment, asshown in FIG. 5 a . So, it is essential to find a way to ensure that theratio of speech to noise received by any remote listener is weighted infavor of speech, as in the waveform shown in FIG. 5 b.

Direct and Reflected Signals

A microphone picking up speech in a room will first pick up the voice—asound which initially arrives at the microphone directly from the personspeaking. This microphone will then pick up a series of additionalsound, as the voice is reflected from the room's walls, ceiling andfloor. Unlike the voice, which is not reflected, these sounds arrive atthe microphone from all directions, as shown in FIG. 6 .

The difference between direct and reflected signals arriving at amicrophone are a question of time and strength—reflected sounds arrivelater than direct sounds (at different ‘phase’), and they arrive withless energy (or ‘amplitude’).

When a microphone receives direct and reflected sound at a similarlevel, the effect is a ‘blurring’ of the signal conveyed to the remotelistener. This affects the intelligibility of speech. For small meetingroom (2-6 persons), this is perceived by the remote listener as if thespeaker was standing in a bathroom. The waveform shown in FIG. 7 arepresent a signal arriving directly at the microphone from the source.The waveform shown in FIG. 7 b is the same signal as shown in FIG. 7 awith the source sound reverberating in the room.

To overcome noise and reverberation in a typical meeting room, the idealspeakerphone would use a microphone more sensitive to the direction ofspeech than to the direction of sources of noise and reverberation. Thisis known as a ‘directional’ microphone.

All microphones are designed with a certain ‘pickup pattern’, determinedhow directional a microphone is to sound arriving from any specificdirection. These vary between omnidirectional patterns, as shown in FIG.8 a , (equally sensitive to sound from any direction) to bi-directional,as shown in FIG. 8 b , (sensitive to sound from two directions). Ofthese pickup patterns, the simplest and most commonly available is theomnidirectional microphone.

Producing Directional Pickup from Omnidirectional Microphones

It is possible to produce a focused and directional pickup of sound froma series of omnidirectional microphones. By configuring thesemicrophones in such a way that differences between the amplitude and thephase of their signals can be manipulated, it is even possible to focusand re-focus the array in any direction, producing the effect of anadaptable directional microphone. This is known as a microphone arraybeamformer. FIG. 9 shows a diagram showing the functioning of a simplebeamforming system known as a ‘Delay and Sum’ beamformer.

The Function of ‘Delay and Sum’ Beamformer

In the ‘Delay and Sum’ beamformer configuration sound arrives at thearray on an angle on the left. On account of its angled arrival, soundreaches the array's microphones at different times. These differences intime are determined by the amount of distance between the microphones.

By introducing specific delays to each microphone, it is possible toalign the signals in such a way so as to synchronize them and to focuson sound coming from a certain direction. The process is known as‘constructive interference’. By adjusting these delays, it is possibleto virtually ‘steer’ the array to ‘focus’ on sound arriving from anyspecific direction. The geometry of the array and the precise amount ofdelay must be carefully designed if the system is to function accuratelyand flexibly.

In the speakerphone of the present disclosure, six low-noise digitalMEMS microphones are placed in a specific configuration optimized toenable speech to be picked up from any angle, relative to thespeakerphone system. Without signal processing, the speakerphone wouldwork as an omnidirectional microphone. When all six microphones areengaged and the advanced signal processing algorithms are applied,however, the directional pattern is focused into a tight beam.

Focusing on Speech

A focused, steerable beam like this is useful for optimizing the targetsound while simultaneously rejecting sound coming from other directions,allowing the pickup of speech from any angle desired, relative to thespeakerphone system. The system is capable of analyzing the content inall directions and automatically selecting the direction of interest.

In FIG. 10 a-d , it can be seen that, even if the target speech signalchanges position, (e.g. when two different people are present in ameeting room), the speakerphone will automatically steer a focused beamto the desired direction for the target signal.

Stationary Noise Reduction

Using a certain number of beams to cover the entire 360° in thehorizontal plane, each individual beam can be tightly focused. Soundsarriving from the same direction that the beam is pointing in will bepicked up without any change, as compared with an omnidirectionalmicrophone. Sounds arriving from other angles, meanwhile, are greatlyattenuated—that is, diminished.

Active De-Reverberation

As we have seen, reverberation causes sounds to arrive at aspeakerphone's microphone with additional delay and from additionalangles. This results in a blurring of the signal in time, reducing theintelligibility of speech.

An optimal ratio of speech to reverberated sound is maintained throughthe use of a beam focused in the direction of the target signal. Soundsarriving at an angle, reflected from the room's surfaces, will beconveyed with attenuation, in comparison to the target signal, as shownin FIG. 11 .

In an aspect, a headset or headphone comprising a loudspeaker system asdescribed above in the detailed description, in the drawings and in theclaims is furthermore provided.

The loudspeaker system according to the present disclosure is in generalapplicable in any device or system comprising a specific electroacousticsystem (e.g. including a loudspeaker and a mechanical parts incommunication with the loudspeaker) with a resulting transfer function,which (at a specific sound output) exhibits a low frequency and/or highfrequency drop-off (as schematically shown for a loudspeaker unit inFIG. 3 c ). The application of a loudspeaker system according to thepresent disclosure may advantageously contribute to a compensation forloss of low and/or high frequency components of the sound output to theenvironment (e.g. due to leakage).

In a headset or headphone, the drop-off is primarily determined by theelectroacoustic system (including A. the loudspeaker design, and B. theear-cup/ear pad/ear bud/earpiece design).

The headphone or headset may be of the open type (indicating a certainor substantial exchange of sound with the environment). The headphone orheadset may be of the closed type (indicating an aim to limit theexchange of sound with the environment).

The term open may in the present context be taken to mean that arelatively high acoustic leakage between the surrounding environment anda volume limited by to the ear/ear canal/ear drum and an ear cup/earpad/ear bud of the headset or head phone covering or blocking theear/ear canal. In a closed type headphone or headset, the leakage willbe (substantially) less than in an open type (but some leakage willtypically be present; and this leakage can be compensated for by aloudspeaker system according to the present disclosure).

The headphone or headset may comprise an active noise cancellationsystem.

In a further aspect, use of a loudspeaker system as described above inthe detailed description below, in the drawings. Use in a speakerphone,or a mobile (e.g. cellular) telephone (e.g. a SmartPhone), or a gamingdevice, or a headphone, or a headset, or a hearing aid is provided.

The communication device may be portable device, e.g. a devicecomprising a local energy source, e.g. a battery, e.g. a rechargeablebattery. The hearing assistance device may be a low power device. Theterm ‘low power device’ is in the present context taken to mean a devicewhose energy budget is restricted, e.g. because it is a portable device,e.g. comprising an energy source, which—without being exchanged orrecharged—is of limited duration (the limited duration being e.g. of theorder of hours or days).

The communication devices may comprise an analogue-to-digital converter(ADC) to digitize an analogue input with a predefined sampling rate,e.g. 20 kHz. The communication device may comprise a digital-to-analogueconverter (DAC) to convert a digital signal to an analogue outputsignal, e.g. for being presented to a user or users via an outputtransducer.

The frequency range considered by the communication device may be from aminimum frequency f_(min) to a maximum frequency f_(max) and maycomprise a part of the typical human audible frequency range from 20 Hzto 20 kHz, e.g. a part of the range from 20 Hz to 12 kHz.

In particular, the communication device may comprise a voice detector(VD) for determining whether or not an input signal comprises a voicesignal (at a given point in time). Such detector may aid in determiningan appropriate mode of operation of the loudspeaker system

The communication device may comprise an acoustic (and/or mechanical)feedback suppression system. The communication device may furthercomprise other relevant functionality for the application in question,e.g. compression, noise reduction, etc.

The communication device may comprise a cellular telephone or aspeakerphone. The communication device may comprise or be a listeningdevice, e.g. an entertainment device, e.g. a music player, e.g. ahearing aid, e.g. a hearing instrument, e.g. a headset, an earphone, anear protection device or a combination thereof.

When the loudspeaker system are headphones, then the advantage of thepresent disclosure is that an audio signal comprising music, or anysound can be easily distributed between the communication unit.

The audio signal being transmitted via the communication link maycomprise speech or speeches from one or more users of another audiogateway connected to the other end of the first communication link.Alternatively, the audio signal may comprise music. In this example, themusic may then be played on the communication units simultaneously.

A hearing device may be or include a hearing aid that is adapted toimprove or augment the hearing capability of a user by receiving anacoustic signal from a user's surroundings, generating a correspondingaudio signal, possibly modifying the audio signal and providing thepossibly modified audio signal as an audible signal to at least one ofthe user's ears. The “hearing device” may further refer to a device suchas a hearable, an earphone or a headset adapted to receive an audiosignal electronically, possibly modifying the audio signal and providingthe possibly modified audio signals as an audible signal to at least oneof the user's ears. Such audible signals may be provided in the form ofan acoustic signal radiated into the user's outer ear, or an acousticsignal transferred as mechanical vibrations to the user's inner earsthrough bone structure of the user's head and/or through parts of middleear of the user or electric signals transferred directly or indirectlyto cochlear nerve and/or to auditory cortex of the user.

The hearing device is adapted to be worn in any known way. This mayinclude arranging a unit of the hearing device behind the ear with atube leading air-borne acoustic signals into the ear canal or with areceiver/loudspeaker arranged close to or in the ear canal such as in aBehind-the-Ear type hearing aid, and/or arranging the hearing deviceentirely or partly in the pinna and/or in the ear canal of the user suchas in an In-the-Ear type hearing aid or In-the-Canal/Completely-in-Canaltype hearing aid, or arranging a unit of the hearing device attached toa fixture implanted into the skull bone such as in Bone Anchored HearingAid or Cochlear Implant, or arranging a unit of the hearing device as anentirely or partly implanted unit such as in Bone Anchored Hearing Aidor Cochlear Implant.

A “hearing system” refers to a system comprising one or two hearingdevices, and a “binaural hearing system” refers to a system comprisingtwo hearing devices where the devices are adapted to cooperativelyprovide audible signals to both of the user's ears. The hearing systemor binaural hearing system may further include auxiliary device(s) thatcommunicates with at least one hearing device, the auxiliary deviceaffecting the operation of the hearing devices and/or benefitting fromthe functioning of the hearing devices. A wired or wirelesscommunication link between the at least one hearing device and theauxiliary device is established that allows for exchanging information(e.g. control and status signals, possibly audio signals) between the atleast one hearing device and the auxiliary device. Such auxiliarydevices may include at least one of remote controls, remote microphones,audio gateway devices, mobile phones, public-address systems, car audiosystems or music players or a combination thereof. The audio gateway isadapted to receive a multitude of audio signals such as from anentertainment device like a TV or a music player, a telephone apparatuslike a mobile telephone or a computer, a PC. The audio gateway isfurther adapted to select and/or combine an appropriate one of thereceived audio signals (or combination of signals) for transmission tothe at least one hearing device. The remote control is adapted tocontrol functionality and operation of the at least one hearing devices.The function of the remote control may be implemented in a Smartphone orother electronic device, the Smartphone/electronic device possiblyrunning an application that controls functionality of the at least onehearing device.

In general, a hearing device includes an input unit such as a microphonefor receiving an acoustic signal from a user's surroundings andproviding a corresponding input audio signal, and/or a receiving unitfor electronically receiving an input audio signal.

The hearing device further includes a signal processing unit forprocessing the input audio signal and an output unit for providing anaudible signal to the user in dependence on the processed audio signal.

The input unit may include multiple input microphones, e.g. forproviding direction-dependent audio signal processing. Such directionalmicrophone system is adapted to enhance a target acoustic source among amultitude of acoustic sources in the user's environment. In one aspect,the directional system is adapted to detect (such as adaptively detect)from which direction a particular part of the microphone signaloriginates. This may be achieved by using conventionally known methods.The signal processing unit may include amplifier that is adapted toapply a frequency dependent gain to the input audio signal. The signalprocessing unit may further be adapted to provide other relevantfunctionality such as compression, noise reduction, etc. The output unitmay include an output transducer such as a loudspeaker/receiver forproviding an air-borne acoustic signal transcutaneous or percutaneous tothe skull bone or a vibrator for providing a structure-borne orliquid-borne acoustic signal. In some hearing devices, the output unitmay include one or more output electrodes for providing the electricsignals such as in a Cochlear Implant.

Further objects of the application are achieved as defined in thedependent claims and in the detailed description of the disclosure.

BRIEF DESCRIPTION OF DRAWINGS

The aspects of the disclosure may be best understood from the followingdetailed description taken in conjunction with the accompanying figures.The figures are schematic and simplified for clarity, and they just showdetails to improve the understanding of the claims, while other detailsare left out. Throughout, the same reference numerals are used foridentical or corresponding parts. The individual features of each aspectmay each be combined with any or all features of the other aspects.These and other aspects, features and/or technical effect will beapparent from and elucidated with reference to the illustrationsdescribed hereinafter in which:

FIG. 1 shows a hearing device/loudspeaker system according to thepresent disclosure, and

FIG. 2 shows room properties for a loudspeaker system according to thepresent disclosure.

FIG. 3 shows a simplified block diagram for optimizing microphonesignals in a room according to the present disclosure,

FIG. 4 a-b shows two application scenarios of a speakerphone or aheadset comprising a loudspeaker system according to the presentdisclosure,

FIG. 5 a shows an exemplary waveform of unintelligible speech,

FIG. 5 b shows an exemplary waveform of clean speech,

FIG. 6 shows exemplary direct & reflected signals arriving at amicrophone,

FIG. 7 a shows an exemplary waveform of direct signal,

FIG. 7 b shows an exemplary waveform of direct signal, as shown in FIG.7 a , & reverberation signals,

FIG. 8 a shows an omnidirectional microphone pickup pattern,

FIG. 8 b shows a bidirectional microphone pickup pattern,

FIG. 9 shows a ‘Delay & Sum’ beamformer system,

FIG. 10 a shows an omnidirectional pickup pattern,

FIG. 10 b shows a focused beam pickup pattern,

FIG. 10 c shows a focused beam steered to 0°,

FIG. 10 d shows a focused beam steered to 210°,

FIG. 11 shows a focused beam pickup pattern.

Further scope of applicability of the present disclosure will becomeapparent from the detailed description given hereinafter. However, itshould be understood that the detailed description and specificexamples, while indicating preferred embodiments of the disclosure, aregiven by way of illustration only. Other embodiments may become apparentto those skilled in the art from the following detailed description.

The detailed description set forth below in connection with the appendeddrawings is intended as a description of various configurations. Thedetailed description includes specific details for the purpose ofproviding a thorough understanding of various concepts. However, it willbe apparent to those skilled in the art that these concepts may bepracticed without these specific details. Several aspects of theapparatus and methods are described by various blocks, functional units,modules, components, circuits, steps, processes, algorithms, etc.(collectively referred to as “elements”). Depending upon particularapplication, design constraints or other reasons, these elements may beimplemented using electronic hardware, computer program, or anycombination thereof.

The electronic hardware may include micro-electronic-mechanical systems(MEMS), integrated circuits (e.g. application specific),microprocessors, microcontrollers, digital signal processors (DSPs),field programmable gate arrays (FPGAs), programmable logic devices(PLDs), gated logic, discrete hardware circuits, printed circuit boards(PCB) (e.g. flexible PCBs), and other suitable hardware configured toperform the various functionality described throughout this disclosure,e.g. sensors, e.g. for sensing and/or registering physical properties ofthe environment, the device, the user, etc. Computer program shall beconstrued broadly to mean instructions, instruction sets, code, codesegments, program code, programs, subprograms, software modules,applications, software applications, software packages, routines,subroutines, objects, executables, threads of execution, procedures,functions, etc., whether referred to as software, firmware, middleware,microcode, hardware description language, or otherwise.

In general, a hearing device includes i) an input unit such as amicrophone for receiving an acoustic signal from a user's surroundingsand/or the user's own voice and providing a corresponding input audiosignal, and/or ii) a receiving unit for electronically receiving aninput audio signal. The hearing device further includes a signalprocessing unit for processing the input audio signal and an output unitfor providing an audible signal to the user in dependence on theprocessed audio signal.

FIG. 1 shows a loudspeaker system 10 an input unit IU providing anelectric input audio signal eIN based on input signal IN. Input signalIN may e.g. be an acoustic signal from the environment (in which caseinput unit IU comprises a microphone), or an electric signal receivedfrom a component of the loudspeaker system or from another device, or amixture thereof. The input unit IU comprises an audio interface. Theinput signal IN may (in case it is an electric signal) be an analoguesignal (e.g. an audio signal from an audio jack interface) or a digitalsignal (e.g. an audio signal from a USB audio interface). The input unit11 (IU) may e.g. comprise an analogue to digital converter (ADC) toconvert an analogue electric signal to a digital electric signal (usingan appropriate audio sampling frequency, e.g. 20 kHz). The loudspeakersystem 10 further comprises a processing unit 13 for modifying theelectric input audio signal 12 (eIN) (or a processed version thereof)according to a predefined optimization function 100. The loudspeakersystem 10 further comprises a loudspeaker unit 15 (SPK) for convertingthe optimized electric audio signal 14 (eINeq) to an acoustic outputsound signal OUT. The loudspeaker unit 14 (SPK) may alternatively be amechanical vibrator of a bone anchored hearing device. In a particularmode of operation of the loudspeaker system 10, the processing unit 13is configured to apply a specific optimization function to the electricinput audio signal.

FIG. 2 shows different room properties in which the microphone system isused. In the room in FIG. 2 a , the room is a low-reverberation room,and the mixing ratio is chosen as 1:0, i.e. all omnidirectional input.In the room for FIG. 2 b , the room is a highly reverberant room, andthe mixing ratio is chosen to be 0:1, ie. all directional so as to focusmaximally towards the source. In the room for 2c, the room is a mediumreverberant room where the mixing ratio is chosen to be 0.5:0.5, i.e. a50% mix of directional and omnidirectional input. This yields a combineddirectional pattern as outlined in the figure. In the room for FIG. 2 d, the room is low-medium reverberant, i.e. not as reverberant as room 2c, but more than in FIG. 2 a . Here the mixing ratio is set to0.67:0.33, i.e. more of the omnidirectional signal is used than thedirectional signa, yields a slightly more round directional pattern asshown in FIG. 2 d . As indicated in FIG. 2 d , two or more digits may bedesirable. Here the omnidirectional signal will be roughly weighted with⅔ and the directional signal will be weighted with ⅓ and would fit alow-medium reverb room.

FIG. 3 shows a method S100 for optimizing the microphone signalperformance based on room properties as shown in FIG. 2 . The methodcomprising the steps of receiving S10 acoustic input signals 12,processing S20 said acoustic input signals by using an algorithm 100 foroptimizing the microphone signal performance and transmitting S30 andoutput signal 13 based on said processed input signal.

The processing further comprises steps of mixing S21 the beams with anomni response to some ratio determined by the room properties.

-   -   1. In a hard room maximum directionality is wanted. This is        achieved by mixing omni/beam in a close to 0:1 relationship.    -   2. In a very soft room the full (or close to) omni directional        response is preferred. This is achieved by an opposite mix of        1:0.    -   3. In rooms between these extremes the mixture will be in        between.

One way of achieving this is to assess the coefficients in an echocanceller (also present in the device) filter as they stronglycorrelates with the room reverb and use this information to select a mixbetween e.g. directional filters and omni directional pickup patternhaving the optimal tradeoff between reverb and noise.

By using an array of microphones both a directional, omni and enhancedomni system can be realized.

A concrete example could be to use a microphone array with 6 microphonesto generate 6 pickup beams and one enhanced omnidirectional “beam”.

FIG. 4 a shows a communication device CD comprising two wired orwireless audio interfaces to other devices, a) a wireless telephone(CellPh, e.g. a cellphone, e.g. a Smartphone, FIG. 4 a ) or a one-wayaudio delivery device (Music player in FIG. 4 b ), and b) a computer(PC, e.g. a PC). The audio interface to the computer (PC) maybe wirelessor a wired, e.g. an USB (audio) interface including a cable and anUSB-connector, for connecting the communication device to the computerand allowing two-way audio to be exchanged between the communicationdevice CD and the computer. The audio interface to the wirelesstelephone (CellPh) may comprise a cable and/or a phone connector (PhCon)for directly connecting the communication device to the computer andallowing two-way audio to be exchanged between the communication deviceand the computer. The communication device (CD) comprises a number ofactivation elements (B1, B2, B3), e.g. push buttons (or alternatively atouch sensitive display) allowing the control of functions of thecommunication device and/or devices connected to the communicationdevice. One of the activation elements (e.g. B1) may be configured toallow connection (hook-off, answer call) and/or dis-connection (hook-on,terminate call) of the wireless telephone (CellPh) connected to thecommunication device. One of the activation elements (e.g. B2) may beconfigured to allow a user to control the volume of the loudspeakeroutput. One of the activation elements (e.g. B3) may be configured toallow a user to control a mode of operation of the loudspeaker system ofthe communication device.

The scenario shown in FIG. 4 a illustrates a teleconference betweenusers (U1, U2) in the vicinity of the communication device (CD) andusers (RU1, RU2, and RU3) at two (different) remote locations. Remoteuser RU1 is connected to the communication device (CD) via wirelesstelephone (CellPh) and wireless connection WL1 to a network (NET).Remote users RU2, RU3 are connected to the communication device (CD) viacomputer (PC) and wired connection WI1 to a network (NET).

FIG. 4 b illustrates a different scenario than FIG. 4 a . FIG. 4 billustrates the reception (and optional mixing) of audio signals fromthe various audio delivery devices (Music player and PC) connected tothe communication device (CD). The communication device (CD) comprisestwo (two-way) audio interfaces embodied in I/O units IU1/OU1 andIU2/OU2, respectively.

The communication device of FIG. 4 b comprises a loudspeaker signal path(SSP), a microphone signal path (MSP), and a control unit (CONT) fordynamically controlling signal processing of the two signal paths. Theloudspeaker signal path (SSP) comprises receiver units (IU1, IU2) forreceiving an electric signal from a connected device and providing it asan electric received input signal (S-IN1, S-IN2), an SSP-signalprocessing unit 13 a for processing (including equalizing) the electricreceived input signal (S-IN1, S-IN2) and providing a processed outputsignal (S-OUT), and a loudspeaker unit 15 operationally connected toeach other and configured to convert the processed output signal (S-OUT)to an acoustic sound signal (OS) originating from the signal received bythe receiver unit (IU1, IU2). The loudspeaker signal path (SSP) furthercomprises a selector-mixing unit (SEL-MIX) for selecting one of the twoinputs audio signals (or mixing them) and providing a resulting signalS-IN to the SSP-signal processing unit 13 a. The microphone signal path(MSP) comprises a microphone unit (MIC) for converting an acoustic inputsound (IS) to an electric microphone input signal (M-IN), an MSP-signalprocessing unit 13 b for processing the electric microphone input signal(M-IN) and providing a processed output signal (M-OUT), and respectivetransmitter units (OU1, OU2) operationally connected to each other andconfigured to transmit the processed signal (M-OUT) originating from aninput sound (IS) picked up by the microphone unit (MIC) to the connecteddevice. The control unit (CONT) is configured to dynamically control theprocessing of the SSP- and MSP-signal processing units 13 a and 13 b,respectively, including mode selection, and equalization in the SSPpath.

The loudspeaker signal path (SSP) is divided in two (IU1, IU2) forreceiving input signals from the respective audio devices (Music playerand PC). Likewise, the microphone signal path (MSP) is divided in two(OU1, OU2) for transmitting output signals to the respective audiodevices (Music player (not relevant) and PC). One-way and two-way audioconnections between the communication device (units IU1, IU2 and OU1,OU2) and two the audio devices (here Music player and PC) can beestablished via wired or wireless connections, respectively.

It is intended that the structural features of the devices describedabove, either in the detailed description and/or in the claims, may becombined with steps of the method, when appropriately substituted by acorresponding process.

As used, the singular forms “a,” “an,” and “the” are intended to includethe plural forms as well (i.e. to have the meaning “at least one”),unless expressly stated otherwise. It will be further understood thatthe terms “includes,” “comprises,” “including,” and/or “comprising,”when used in this specification, specify the presence of statedfeatures, integers, steps, operations, elements, and/or components, butdo not preclude the presence or addition of one or more other features,integers, steps, operations, elements, components, and/or groupsthereof. It will also be understood that when an element is referred toas being “connected” or “coupled” to another element, it can be directlyconnected or coupled to the other element, but an intervening elementmay also be present, unless expressly stated otherwise. Furthermore,“connected” or “coupled” as used herein may include wirelessly connectedor coupled. As used herein, the term “and/or” includes any and allcombinations of one or more of the associated listed items. The steps ofany disclosed method is not limited to the exact order stated herein,unless expressly stated otherwise.

It should be appreciated that reference throughout this specification to“one embodiment” or “an embodiment” or “an aspect” or features includedas “may” means that a particular feature, structure or characteristicdescribed in connection with the embodiment is included in at least oneembodiment of the disclosure. Furthermore, the particular features,structures or characteristics may be combined as suitable in one or moreembodiments of the disclosure. The previous description is provided toenable any person skilled in the art to practice the various aspectsdescribed herein. Various modifications to these aspects will be readilyapparent to those skilled in the art, and the generic principles definedherein may be applied to other aspects.

The claims are not intended to be limited to the aspects shown hereinbut are to be accorded the full scope consistent with the language ofthe claims, wherein reference to an element in the singular is notintended to mean “one and only one” unless specifically so stated, butrather “one or more.” Unless specifically stated otherwise, the term“some” refers to one or more.

Accordingly, the scope should be judged in terms of the claims thatfollow.

1. A method for optimizing speech pickup in a communication deviceutilizing a loudspeaker system, wherein the communication devicecomprises a microphone system placed in a specific configuration,wherein the method comprising: receiving acoustic input signals by themicrophone system, providing a directional output signal based onsignals from said microphone system, providing an omnidirectional outputsignal based on signals from said microphone system, processing saidacoustic input signals by determining a mixing ratio according to adetermined reverberation level in a room in which the microphone systemis placed, and mixing said directional output sound signal with saidomnidirectional output signal based on the determined mixing ratio, themixing ratio defining a ratio of omnidirectional output signal to thedirectional output signal, and transmitting an output signal based onsaid processing.
 2. The method according to claim 1, wherein themicrophone system comprises at least six microphones.
 3. The methodaccording to claim 2, wherein the microphones are low-noise digital MEMSmicrophones are placed in a specific configuration optimized to enablespeech to be picked up from any angle relative to the communicationdevice.
 4. The method according to claim 1, wherein the microphonesystem is engaged, and the advanced signal processing algorithms areapplied, and the directional pattern is focused into a tight beam. 5.The method according to claim 1, wherein the method further comprisesanalyzing the content of the received sound signal in all directions,and automatically selecting the direction of interest, the target soundsignal.
 6. The method according to claim 1, wherein the communicationdevice is configured to steer a focused beam to the desired directionfor the target signal, even if the target sound signal changes position.7. The method according to claim 1, further comprising: determining thereverberation level in the room using a reverb estimator.
 8. The methodaccording to claim 7, wherein the method comprising continuouslyupdating the mixing ratio based on the determined reverberation level inthe room.
 9. A system for optimizing speech pickup in aloudspeaker-based communication device of the system, comprising: aninput unit comprising a microphone system placed in a specificconfiguration and adapted for receiving audio input signals andproviding electric audio input signals; a mixing unit for determining amixing ratio based on a determined reverberation level in a room inwhich the input unit is placed, and mixing on the basis of thedetermined mixing ratio a directional signal obtained from said electricaudio input signals with an omnidirectional signal obtained from saidelectric audio input signals, and an output unit for providing an outputsignal perceivable by a user as sound representative of said electricaudio input signals or a processed version thereof.
 10. The systemaccording to claim 9, wherein the microphone system comprises at leastsix microphones.
 11. The system according to claim 10, wherein themicrophones are low-noise digital MEMS microphones are placed in aspecific configuration optimized to enable speech to be picked up fromany angle relative to the communication device.
 12. The system accordingto claim 9, wherein a processing unit is configured for analyzing thecontent of the received sound signal in all directions, andautomatically selecting a direction of interest comprising a targetsound signal.
 13. The system according to claim 10, wherein thecommunication device is adapted to automatically adaptively steer afocused beam to a desired direction
 14. The system according to claim10, wherein the mixing ratio is continuously updated.
 15. Aloudspeaker-based communication device for optimizing speech pickup insaid communication device, wherein the communication device comprises:an input unit comprising a microphone system placed in a specificconfiguration and adapted for receiving audio input signals andproviding electric audio input signals; a mixing unit for determining amixing ratio based on a determined reverberation level in a room inwhich the input unit is placed, and mixing on the basis of thedetermined mixing ratio a directional sound signal obtained from saidelectric audio input signals with an omnidirectional signal obtainedfrom electric audio input signals, and an output unit for providing anoutput signal perceivable by a user as sound representative of saidelectric audio input signals or a processed version thereof.
 16. Thecommunication device according to claim 15, wherein the microphonesystem comprises at least six microphones.
 17. The communication deviceaccording to claim 16, wherein the microphones are low-noise digitalMEMS microphones are placed in a specific configuration optimized toenable speech to be picked up from any angle.
 18. The communicationdevice according to claim 15, wherein a processing unit is adapted foranalyzing the content of the received sound signal in all directions,and automatically selecting the direction of interest, and providing anadaptive directional beamformed signal.
 19. The communication deviceaccording to claim 15, wherein the speakerphone system is adapted toautomatically steer a focused beam adaptively to a desired direction.20. The communication device according to claim 15, wherein the mixingratio is continuously updated.